1. Field of the Invention
The present invention relates to providing subscriber information in a Voice over IP (VoIP) system and, more particularly, to a method and apparatus of providing subscriber information in a VoIP system capable of providing, as subscriber information, a variety of multimedia information instead of simple text-based subscriber information when providing a caller identification service.
2. Description of the Related Art
The Voice over IP (VoIP) refers to an Internet telephony technology for a series of apparatus that transfers voice information using an internet protocol (IP). The VoIP is not a line based protocol like a protocol for a public switched telephone network (PSTN). The VoIP transfers voice information in a digital form within discrete packets.
This VoIP or Internet telephony technology enables telephone users to receive toll and international call services only with a local telephone rate under an Internet or Intranet environment by utilizing an existing IP network as it is to implement an integrated voice call service.
For example, H.323, a session initiation protocol (SIP), and a media gateway control protocol (MGCP) have been defined as protocols for implementing the VoIP. At present, the SIP, which is a simple text-based application layer control protocol, is widely being used.
The SIP is intended to establish and control sessions between terminals or users. The SIP is a simple text-based application layer control protocol and enables one or more participants to establish, modify and terminate the sessions in a cooperative manner. Such sessions can include, for example, Internet teleconferencing, telephony, meeting, and instant messaging services.
The SIP was developed by the Multiparty Multimedia Session Control (MMUSIC) working group of the Internet Engineering Task Force (IETF), and was published in March, 1993. Its standard draft was generally described in ‘Request for Comments (RFC) 3261.’
To make a voice call to a called IP terminal, a calling IP terminal transmits an ‘INVITE message’, which is a call request message, to the called IP terminal.
The called IP terminal transmits a return message in response to the call request message received from the calling IP terminal to setup a call with the calling IP terminal.
The calling IP terminal transmits the ‘INVITE’ message as the call request message to the called IP terminal.
The called IP terminal, which has received the ‘INVITE’ message, transmits a ‘180 ringing’ message to the calling IP terminal to notify the calling IP terminal that the called IP terminal has received the ‘INVITE’ message.
The called IP terminal also transmits a ‘200 OK’ message in response to the ‘INVITE’ message to notify the calling IP terminal that the called IP terminal has accepted the invitation to the session.
In response to the ‘200 OK’ message, the calling IP terminal transmits an ‘ACK’ message, which is a final acknowledgment message, to the called IP terminal. This sets up a call between the calling IP terminal and the called IP terminal to perform a voice call.
When the called IP terminal desires to terminate the call after the call has been established and the voice call has been performed between the calling IP terminal and the called IP terminal as described above, the called IP terminal transmits a ‘BYE’ message to the calling IP terminal in order to terminate the call.
The calling IP terminal transmits the ‘200 OK’ message to the called IP terminal to acknowledge the ‘BYE’ message after receiving the ‘BYE’ message from the called IP terminal, resulting in a call termination between the IP terminals.
In a typical VoIP based voice call service, subscriber information including Caller IDentification (CID) information of the calling IP terminal is transferred to the called IP terminal by transferring simple text-based CID information of the calling IP terminal to the called IP terminal.